| 00:00.05 | budmang714 | Ive done this tons before(im using a virtualized openvz enviorment), I used to use zaptel.. |
| 00:00.11 | budmang714 | I think I might have figured it out. |
| 00:05.20 | budmang714 | Bleh, had to copy the user.h rookie mistake. |
| 00:11.21 | *** join/#asterisk lmsteffan (~lmsteffan@reef.ac-noumea.nc) |
| 00:12.26 | *** join/#asterisk luar (~luar@cyberplant-1-pt.tunnel.tserv13.ash1.ipv6.he.net) |
| 00:16.13 | *** join/#asterisk Imo (~Imo@p4FC5CA3A.dip0.t-ipconnect.de) |
| 00:23.38 | *** join/#asterisk Imo (~imox1234@p4FC5CA3A.dip0.t-ipconnect.de) |
| 00:24.11 | Imo | hello, i have an dtmf problem can somebody help me ? |
| 00:26.26 | devoid | Imo: Like what? |
| 00:26.43 | Imo | i cant sent dtmf key |
| 00:26.50 | Imo | i dont understand this |
| 00:27.18 | Imo | i have the same server with the same system on another server. and there are works |
| 00:27.35 | *** join/#asterisk Freeaqingme (~dolf@2001:888:1b7e:0:b0fe:78ff:feb3:1148) |
| 00:28.10 | Freeaqingme | I'm atm looking at buying a Digium TDM400P, is that still a good thing to do, or is there some better alternative (same/better priced)? |
| 00:28.24 | Imo | have you any idea ? |
| 00:28.40 | Freeaqingme | me? |
| 00:28.47 | Imo | no |
| 00:28.50 | Freeaqingme | kk |
| 00:28.54 | Imo | devoid :) |
| 00:29.29 | ManxPower-work | I happen to prefer Sangoma over Digium, but I also don't like analog cards |
| 00:29.35 | devoid | What type of line are you using? |
| 00:30.24 | Imo | you mean me ? |
| 00:36.26 | Imo | when i call intern i can sent dtmf key's please help me ? |
| 00:37.41 | *** join/#asterisk kolby (~kolby@117.104.128.214) |
| 00:41.03 | Imo | ??? |
| 00:42.27 | kolby | can Asterisk be used to make a dail-up internet service on a server? |
| 00:45.46 | Imo | ?????? |
| 00:45.57 | Imo | devoid please help me |
| 00:47.00 | devoid | Imo: What type of line are you using? |
| 00:48.04 | Imo | sorry for my question what what du you mean with line ? |
| 00:53.11 | Imo | ?? |
| 00:54.08 | devoid | Like Zap, DAHDI, SIP, et cetera. |
| 00:54.50 | Imo | SIP |
| 00:58.10 | *** join/#asterisk Taco24502 (~Taco24502@c-67-187-29-41.hsd1.va.comcast.net) |
| 00:59.12 | Imo | dtmf dont work with outgoing calls ? |
| 00:59.21 | Imo | please help me 5 min. or so |
| 00:59.53 | Imo | please let me know where can i find an error ? |
| 01:00.02 | Imo | i have a new install from asterisk |
| 01:00.07 | Imo | but the same problem |
| 01:02.23 | *** join/#asterisk pigo (~7c7cf78c@gateway/web/freenode/x-mrdkcvpkztwsnkxy) |
| 01:02.29 | pigo | hi everybody |
| 01:02.41 | Taco24502 | hello |
| 01:02.48 | pigo | hello taco |
| 01:02.57 | pigo | how are you ? |
| 01:03.20 | Taco24502 | good |
| 01:03.51 | pigo | i am looking here some idea on IVR implementation behind asterisk |
| 01:04.07 | pigo | have you done any ? |
| 01:04.33 | Taco24502 | no im a newb |
| 01:05.31 | Taco24502 | any one o there use a asterisk in a Answering Service |
| 01:06.13 | pigo | i have done a very basic level stuff |
| 01:06.18 | pigo | what is your issue |
| 01:06.24 | pigo | ? |
| 01:06.48 | ManxPower-work | ~answers |
| 01:06.49 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 01:07.05 | ManxPower-work | look at the wiki, just remember many of their examples are outdated |
| 01:08.10 | Taco24502 | dont have one just try to feel my way around |
| 01:11.06 | pigo | well, i am trying to record and save a message, but when i invoke my jsp script a voice message pops "A serious error has occured, Exiting" |
| 01:11.18 | pigo | any body haaving any ideas |
| 01:13.06 | ManxPower-work | pigo, sounds like you are using FreePBX |
| 01:13.34 | ManxPower-work | There will be no voice message unless you configured to to play a voice message |
| 01:13.39 | *** join/#asterisk mrbnet_ (~ryanbantz@c-75-73-142-28.hsd1.mn.comcast.net) |
| 01:13.41 | ManxPower-work | ~freepbx |
| 01:13.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:13.47 | ManxPower-work | ~book |
| 01:13.48 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:13.48 | pigo | i am using Asterisk 1.4.29.1 |
| 01:13.57 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:15.36 | pigo | @manx-power it is able to record and playback messages, but when in that vxml script (which does the recording and playback) i also submit my record element to a jsp script so as to save it permanentaly on disk, the ERROR occurs |
| 01:17.04 | mrbnet_ | Will the soundpoint 421 download config files from a user defined site or must it be assigned through dhcp? |
| 01:23.46 | *** join/#asterisk pigo (~7c7cf78c@gateway/web/freenode/x-bmlxanygkywxtmey) |
| 01:27.35 | *** join/#asterisk bmg505 (~leon@196-209-71-68-rndf-esr-3.dynamic.isadsl.co.za) |
| 01:31.50 | Freeaqingme | Assuming I have some voip accounts. Am I right in saying that I let asterisk connect to those voip accounts, and then have my ata devices connect to asterisk? |
| 01:42.35 | *** join/#asterisk KMiLo (~GeniuS@201.228.187.228) |
| 01:44.36 | ManxPower-work | Freeaqingme, Correct |
| 01:44.38 | ManxPower-work | ~book |
| 01:44.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:44.50 | ManxPower-work | mrbnet_, either |
| 01:45.13 | Freeaqingme | ManxPower-work, i'm reading _a lot_, just wanted to know if I'm on the right way |
| 01:45.15 | ManxPower-work | usually onsite are assigned by DHCP, remote phones would have a user defined server |
| 01:45.27 | ManxPower-work | Freeaqingme, your concept is correct. |
| 01:45.47 | Freeaqingme | ManxPower-work, and comms between asterisk and ATA devices is SIP? |
| 01:50.08 | Freeaqingme | speaking of bad timing :'( |
| 01:58.17 | *** join/#asterisk voipmonk (~shido6@ip67-152-80-228.z80-152-67.customer.algx.net) |
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| 02:14.04 | prometheanfire | Taco24502: you were asking about asterisk in a large call center environment in #gentoo-voip? |
| 02:14.20 | Taco24502 | yes |
| 02:15.58 | prometheanfire | I do, but you should really define large |
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| 02:17.47 | Taco24502 | 10 seat |
| 02:18.08 | Taco24502 | about 4000 calls a day |
| 02:18.21 | prometheanfire | ya, we do that (and larger) |
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| 02:19.18 | Taco24502 | we a Answering Service |
| 02:19.49 | prometheanfire | ya, we do inbound and outbound dialing |
| 02:20.10 | Taco24502 | what kind on software do you use |
| 02:20.32 | carrar | Asterisk! |
| 02:20.47 | prometheanfire | well, we use aheeva if outbound is involved (ansering machine detection and the like) |
| 02:20.55 | prometheanfire | ya, it's all asterisk on the backend |
| 02:21.05 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:21.09 | prometheanfire | quemetrics is good for incoming only |
| 02:21.40 | Taco24502 | aheeva i seen that but never ask for ta price |
| 02:21.52 | prometheanfire | I don't know the price |
| 02:22.05 | prometheanfire | I try to stay as far away from money as I can |
| 02:22.38 | Taco24502 | got you on that |
| 02:23.26 | Taco24502 | im looking for some thing to Answer call > Take Message > Deliver said Message |
| 02:23.59 | prometheanfire | you could do that with straight asterisk |
| 02:24.52 | carrar | thats basic voicemail |
| 02:25.00 | prometheanfire | ya |
| 02:25.07 | Taco24502 | well we need a person a answer and then do it all |
| 02:25.31 | prometheanfire | you can still do that with basic asterisk |
| 02:26.48 | Taco24502 | i need a inther face to do it the message can be a fax,email,page,or a call |
| 02:27.17 | Taco24502 | soming thing like this http://www.nsolve.com/ncall.html |
| 02:27.56 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 02:28.03 | prometheanfire | ya, that'll have to be custom methinks |
| 02:31.06 | Taco24502 | thanks |
| 02:33.28 | *** part/#asterisk prometheanfire (~mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net) |
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| 03:22.07 | adr3nalin3 | hey guys how do I disable all of the dahdi hardware that I am not using so it doesn't load at boot? |
| 03:22.47 | adr3nalin3 | ahh.../etc/dahdi/modules |
| 03:24.33 | ChannelZ | just don't start them |
| 03:25.37 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 03:25.59 | ChannelZ | So I have attach=yes in my voicemail.conf. Then in my default vm context I have "123 => 4567,Robin S,robin@blah.com,,attach=no" - but * still attaches the VM to the mails it sends. ?? |
| 03:27.27 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 03:38.08 | trentcreek | In the voip-info page I see this for the AMI....Originate: Originate Call (privilege: call,all) NOTE: starting from 1.6: originate,all What is the privilege part??? |
| 03:40.48 | ChannelZ | The privileges the AMI user needs to use that command |
| 03:40.57 | Defraz | SO I have two IAX trunks setup between two asterisk servers and it works great to make calls out of asterisk server 1 it has the PRI on it, and I can make calls via asterisk server 1 on asterisk server 2 but How can I get a DID to route to asterisk server 2 via IAX? |
| 03:41.24 | Defraz | I setup a SIP trunk on 2 and just put exten=> _2082390322,1,Dial(SIP/2082390322@192.41.17.98) and it works |
| 03:41.29 | Defraz | but I rather do it via iax. |
| 03:41.55 | trentcreek | ChannelZ: Then what makes sure it has enough privilege? |
| 03:41.56 | ChannelZ | So do Dial(IAX2/whatever.com/whateverextension) |
| 03:42.06 | ChannelZ | trentcreek: manager.conf |
| 03:42.20 | carrar | ChannelZ, that was a bug |
| 03:42.30 | ChannelZ | eh? |
| 03:42.32 | carrar | https://issues.asterisk.org/view.php?id=16849 |
| 03:42.35 | trentcreek | ChannelZ: thanks |
| 03:42.52 | ChannelZ | carrar: oh thanks |
| 03:42.58 | Defraz | so like exten=> _2082390322,1,Dial(IAX/192.41.17.98/2082390322) |
| 03:43.09 | Defraz | Don't I have to specify a username? |
| 03:43.51 | Defraz | exten=> _2082390322,1,Dial(IAX/bill:pass@192.41.17.98/2082390322) |
| 03:43.59 | Defraz | the one I used on my trunk? |
| 03:43.59 | ChannelZ | You said you had "trunks" setup which I assume means each side registers with the other |
| 03:44.08 | Defraz | yes they do register |
| 03:44.46 | Defraz | so then do I specify my username then instead of the ip? |
| 03:44.53 | ChannelZ | so IAX2/foo/exten where 'foo' is the remote user you have setup in iax.conf |
| 03:45.04 | Defraz | okay got it. |
| 03:45.15 | ChannelZ | by user I mean peer.. entry.. whatever. [foo] |
| 03:45.19 | Defraz | yea |
| 03:47.04 | Defraz | well slick. |
| 03:48.27 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
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| 03:55.17 | adr3nalin3 | Hey guys I am having a problem where long distance calls that I am transferring from a PRI to a cell phone. Local calls work just fine but on long distance calls I see -- Channel 0/1, span 1 got hangup request, cause 16 |
| 03:56.01 | adr3nalin3 | *Where long distance calls are getting dropped when transferring to cell phone...long day |
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| 04:08.24 | Defraz | ChannelZ that worked like a charm |
| 04:10.48 | trentcreek | ChannelZ: yeah ...Thanks to you..helped me understand why my script stopped working in asterisk 1.6 |
| 04:12.20 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 04:33.54 | ChannelZ | Glad you have it working |
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| 04:53.46 | jmcdowell | looks around the room... |
| 04:56.57 | ChannelZ | ducks down behind the trash can |
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| 05:30.39 | sawgood | Hi everyone! .... I have a freshly built CentOS 5.4 box with an image of it (so if I crash things I can re-image in 5-10 minutes) ... I want to install Asterisk 1.6.2.6 from src |
| 05:30.50 | sawgood | Can you suggest any additional RPM packages I should install 1st? |
| 05:31.05 | sawgood | For dependencies, etc. |
| 05:36.09 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 05:38.21 | Katty | hi |
| 05:39.39 | sawgood | Hi |
| 05:39.41 | sawgood | how you been? |
| 05:40.37 | Katty | been better |
| 05:40.46 | Katty | my ear ringing was keeping me awake |
| 05:43.18 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
| 05:43.50 | sawgood | configure: error: no acceptable C compiler found in $PATH |
| 05:43.55 | sawgood | what should I RPM to overcome this? |
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| 05:50.48 | carrar | hi |
| 05:51.08 | carrar | sawgood, not use RPM's!! |
| 05:51.14 | carrar | and install a compiler |
| 05:51.45 | sawgood | right ... yum install is what I meant |
| 05:51.52 | sawgood | I put on kernel-headers |
| 05:52.00 | sawgood | but I was not sure what else I needed for Asterisk |
| 05:53.12 | carrar | centos? |
| 05:53.56 | carrar | At install, install base, core, development-libs, development-tools & kernel-devel |
| 05:54.03 | carrar | should be good to go |
| 05:54.24 | sawgood | I already have a CentOS 5.4 box with an image ... I did not add those things from the start |
| 05:54.26 | *** join/#asterisk fld (~sak.tomas@80.48.158.122) |
| 05:54.38 | sawgood | can I yum them now? |
| 05:54.51 | carrar | yeah |
| 05:54.59 | *** join/#asterisk x86 (~x86@p3m/member/x86) |
| 05:55.19 | ChannelZ | and then belch like a man |
| 05:56.48 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hobbajanalxoxdxr) |
| 05:57.52 | sawgood | since you guys all use the CLI only ... for the most part anyways ... don't you find it 'rough' to work with the dialplan for doing IVR work? |
| 05:58.12 | carrar | I don't |
| 05:58.40 | sawgood | At least I'll be glad I only have to manage sip.conf and extensions.conf |
| 05:58.56 | sawgood | instead of 'fighting' which file goes where in what sub file |
| 05:59.39 | sawgood | The two yum installs I've done so far are (kernel-headers and) (yum groupinstall 'development tools'_ |
| 05:59.46 | sawgood | can you think of any others I might need? |
| 06:00.04 | carrar | yes |
| 06:00.36 | carrar | but I already told you so no need to repeat it |
| 06:01.33 | ChannelZ | I'm Walt Mossberg. Shut up! |
| 06:02.28 | sawgood | is the syntax really, yum install base? |
| 06:03.17 | *** join/#asterisk aimka (~mk@unaffiliated/aimka) |
| 06:03.33 | ChannelZ | I think you already have base or the thing wouldn't be running |
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| 06:25.30 | fld | has anybody know how to check in dialplan how many free license of FFA is free? |
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| 06:29.30 | ChannelZ | hmm I don't but I'd guess the fax application would fail with some sort of return value that you let you know |
| 06:32.23 | *** join/#asterisk kartik (~koolkarti@117.207.82.240) |
| 06:34.31 | mykhyggz | fax show licences |
| 06:35.24 | mykhyggz | Free fax licenses: 1 |
| 06:36.29 | ChannelZ | programmatically though, in the dialplan |
| 06:36.37 | mykhyggz | Total licensed ports: 1 ... how you check in a dialplan, IDK. |
| 06:37.13 | fld | fax show license that i know |
| 06:37.41 | fld | but mayby is some variable? |
| 06:38.24 | fld | i have 5 license and 7 fax machines |
| 06:38.51 | fld | s i must chack before i send or receive fax |
| 06:40.26 | ChannelZ | RE: look up what variables ReceiveFAX sets, it probably will throw an error that you could trap.. and do what I dunno (send a busy signal?) |
| 06:42.01 | fld | yes i want send a busy signal but when i don't have free license FAXOPT(error) is empty :/ |
| 06:44.03 | fld | must i count a 'fax show sessions' ?? |
| 06:44.43 | fld | sorry for my english but i don't use it 8 years |
| 06:44.55 | kaldemar | there's a dozen ways to do that. i'd consider using GROUP functions to check the group count and set the group in all fax extensions. |
| 06:47.35 | fld | kaldemar: that's good idea, thx a lot |
| 06:48.47 | coppice | wouldn't it be easier to just use the FAX solution that doesn't limit you to 5 channels? |
| 06:49.41 | fld | mayby yes but asterisk have problem with faxes |
| 06:50.03 | ChannelZ | Or buy 2 more channel licenses since apparently these fax machines are a key part of the business |
| 06:50.07 | fld | and in company where i work is impossible to try t.37 faxes |
| 06:50.19 | fld | only fax machines :/ |
| 06:50.48 | ChannelZ | well wait.. you have physical fax machines? |
| 06:52.05 | fld | yes 2 more license i may buy but we receives about 400 faxes a day so a lot of them is in this same time to this same number :/ |
| 06:52.12 | fld | yes |
| 06:52.37 | ChannelZ | So I'm missing something, what does FFA have to do with it? |
| 06:52.51 | coppice | If you are receiving on physical FAX machines, why do you need FAX support in Asterisk? |
| 06:53.06 | fld | i receive fax on asterisk and send it to fax machine using t.38 |
| 06:53.34 | ChannelZ | How do you receive calls |
| 06:54.38 | fld | I use ReceiveFax function and when status is SUCCESS i go to agi script who send this fax again but to my fax machine |
| 06:54.57 | ChannelZ | No.. I mean how do you get the fax from the remote end in the first place? |
| 06:54.59 | coppice | now that is an amusingly weird solution |
| 06:55.57 | ChannelZ | I mean presumably these real fax machines are hooked up to a TDM card of some sort, an ATA providing FXS ports to connect the fax machines to.. |
| 06:56.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:58.20 | fld | no all fax machines are connect to ATA like a phone |
| 06:58.45 | ChannelZ | Like I said. |
| 06:58.51 | fld | yes |
| 06:58.57 | ChannelZ | So how does the FAX come into your system in the first place? |
| 06:59.12 | fld | from E1 |
| 06:59.55 | ChannelZ | So why aren't you just Dial()ing the ATA port the physical fax machine is connected to? |
| 07:01.01 | fld | because i don't have a copy in server and a lot of faxes are failed |
| 07:01.56 | fld | i have fax machines in 13 cities in POLAND and only 3 asterisk servers linked dundi |
| 07:04.18 | ChannelZ | hmm ok |
| 07:05.02 | fld | and i use only 2 E1 whit 200 DDI |
| 07:05.09 | fld | number |
| 07:06.53 | fld | but maybe U know same other good solutions for faxes in asterisk |
| 07:06.54 | fld | ?? |
| 07:08.16 | ChannelZ | well not really based on your requirement that you capture them in software first to save them someplace |
| 07:10.33 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 07:10.45 | ChannelZ | SHOWER TIME! |
| 07:10.53 | ChannelZ | (it's like Hammer Time but with less pants) |
| 07:10.58 | fld | that's now problem, i tell my director that's impossible save this faxes |
| 07:11.36 | coppice | fld: buy two more licences, or use the free alternative |
| 07:11.44 | ChannelZ | Why don't you do the dispatching of faxes via call files? |
| 07:12.41 | fld | what does it mean?? |
| 07:12.57 | ChannelZ | Maybe it would deal with the bother of re-queuing failed ones due to no licenses |
| 07:17.37 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:19.02 | fld | no license are not problem, but what i sad i may buy some new license or counting it. but in FFA i have a problem with send to pstn because all is ok when a send to fax number but sometimes i must send a fax to phone and ask for a fax signal so mayby U know some other solution to send a fax to PSTN LINE |
| 07:20.30 | ChannelZ | My understanding of your original question was you needed to know how many licenses were in use because you had fewer licenses than uses for it in simultaneous use |
| 07:21.26 | fld | yes and kaldemar sad that i may use GROUP function to count this and i think that's good idea |
| 07:22.08 | ChannelZ | I just hadn't realized the topic had changed, sorry. |
| 07:22.09 | ChannelZ | Time to poop. |
| 07:23.12 | fld | ok i go to work |
| 07:23.15 | fld | thx a lot |
| 07:30.46 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 07:35.42 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 07:41.07 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 07:41.25 | sun28 | moin |
| 07:43.13 | ChannelZ | yo |
| 07:54.36 | *** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-bglnszacytpygkyp) |
| 07:55.06 | aceio | hi all |
| 07:57.23 | aceio | can someone take a look at me sip.conf and extensions.conf |
| 07:58.08 | aceio | http://pastebin.ca/1854701 |
| 07:58.41 | Gugge | done |
| 07:59.11 | aceio | asterisk cli is genrating the following OTICE[2482]: chan_sip.c:15144 handle_request_invite: Call from '1260' to extension '1261' rejected because extension not found. |
| 08:00.01 | aceio | can someone explain to me where i am going wrong |
| 08:00.26 | Gugge | theres is no "phones" context |
| 08:00.46 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 08:00.57 | Gugge | so its perfectly right, it cant find 1261 in the phones context |
| 08:03.08 | aceio | okay |
| 08:05.01 | aceio | so do i need to make the change in sip.conf |
| 08:06.07 | ChannelZ | That, but I also don't see any extensions in any of your other contexts that would even match.. |
| 08:06.18 | Gugge | aceio: that depends on what you want |
| 08:06.36 | Gugge | the context= line in sip.conf should have a context that exists, if you want to be able to place calls |
| 08:06.39 | ChannelZ | except for the one that dials out your Zap channel which is probably not what you want |
| 08:08.39 | aceio | i see |
| 08:09.08 | Gugge | maybe you want context=sip in sip.conf and then the first exten in [sip] should be _1XXX ? |
| 08:09.12 | Gugge | but im just guessing |
| 08:09.15 | ChannelZ | Remember, SIP device names are NOT the same thing as extensions |
| 08:10.10 | aceio | i have a context called dial-ext i will add it too sip.conf |
| 08:10.51 | Gugge | but dial-ext does not match a 4 digit number |
| 08:11.21 | Gugge | _1XX only matches 100-199, not 1261 |
| 08:11.24 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:11.51 | aceio | okay i shall change that alos |
| 08:21.14 | *** join/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 08:21.16 | hurdman | hi |
| 08:21.33 | ChannelZ | HAY! |
| 08:26.42 | hurdman | how can i get the subscriber number instead of the national number into a dialplan var ? |
| 08:27.14 | hurdman | http://asterisk.pastebin.com/SDEHqCj0 <= i have test both of CALLERID var but none is ok |
| 08:37.05 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:42.01 | *** join/#asterisk vlt (~dm@suez.activ-job.com) |
| 08:43.07 | Gugge | hurdman: what is the number you expect to get into a var? |
| 08:45.24 | hurdman | from my example : 0123456789 |
| 08:46.05 | vlt | Hello. I'm trying to understand how T.38 works and what is going in my setup here. Is it correct that an app with T.38 termination must be able to generate the analog signals a remote fax expects to be sent (by calling spandsp for example)? |
| 08:46.21 | hurdman | Gugge: the debug says that it's the subscriber number |
| 08:47.22 | vlt | So, in case the connection to the T.38 aware leg of the call drops there's always something talking the analog fax on the other leg, right? |
| 08:47.39 | vlt | *talking to |
| 08:48.25 | coppice | if you are terminating there will only be one leg |
| 08:49.33 | vlt | coppice: I see. What does a T.38 terminating app like asterisk >= 1.6 do with that call? |
| 08:50.09 | vlt | coppice: Receive and save the fax? |
| 08:50.31 | coppice | receive to a TIFF file or send from a TIFF file |
| 08:52.05 | vlt | coppice: What T.38 mode do I need to connect an analog fax machine to a remote T.38 aware device? |
| 08:52.35 | coppice | you need T.38 gateway, which only exists as patches right now |
| 08:53.03 | hurdman | is-it possible to set what TON of CallerID i want to have into CALLERDI(*) vars ? |
| 08:53.16 | hurdman | for an incomming call ? |
| 08:54.47 | vlt | coppice: The gateway part is more complicated because it needs to handle both the T.38 connection on one leg and act like a real fax machine (using DSP) to the other? |
| 08:55.18 | coppice | yes, but why does the complexity concern you? |
| 08:55.56 | vlt | coppice: I'm trying to understand how ot works and what components I need to make faxing work here. |
| 08:56.08 | vlt | s/ot/it/ |
| 08:57.06 | coppice | well, if you look in the bug tracker there are patches to add T.38 gateway functionality, but Digium has been extremely slow in integrating this functionality |
| 08:59.36 | *** join/#asterisk Preatorian (~preatoria@81.18.242.156) |
| 08:59.43 | Preatorian | Good day all |
| 09:00.20 | hurdman | hi Preatorian |
| 09:01.18 | vlt | coppice: As far as I understood I need the two gateways I already have (a Patton SmartNode and an ATA) and an Asterisk version with T.38 pass-through. |
| 09:02.27 | Preatorian | Did any of you ever got the following "ERROR[21256]: chan_sip.c:9374 register_verify: Peer '2001' is trying to register, but not configured as host=dynamic" that kept looping (causing server load to rise to like 10-15%...) |
| 09:03.27 | vlt | Is there a way to check if two T.38 aware endpoints both connected to an Asterisk 1.4 are really talking T.38 with each other? |
| 09:04.03 | coppice | the most convincing check would be to look at the packets with wireshark |
| 09:04.27 | Preatorian | ^^ Indeed, it will be in the first 2 headers. |
| 09:05.28 | vlt | coppice: Thank you. |
| 09:06.06 | *** join/#asterisk pentanol (~pentanol@77.35.49.164) |
| 09:06.43 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 09:09.07 | Preatorian | Hmm, how can i stop a looping process in asterisk without effectively restarting the server? |
| 09:09.27 | vlt | soft hangup, maybe |
| 09:09.40 | Preatorian | i wish it was a call |
| 09:11.44 | jblack | is it an agi? |
| 09:12.06 | Preatorian | its a line i posted earlier |
| 09:12.12 | Preatorian | seems really obvious what the problem is |
| 09:12.34 | vlt | There's a wifi connection between Asterisk and a SIP gateway. Phone calls work quite well with only a few (but acceptable) dropouts but faxes don't get through. From your experince: How well does T.38 work? Should it be possible to send faxes over the wifi connection? |
| 09:12.35 | Preatorian | however... the ip displayed in the second rule is blacklisted anyway so unable to contact the machine. |
| 09:12.57 | jblack | Oh, so you're beign ddossed by something thats trying to register. again and again and again |
| 09:12.59 | Preatorian | Yet, still it keeps looping in asterisk. |
| 09:13.19 | jblack | I would limit new connection attempts for that ip to X per min. |
| 09:13.25 | jblack | in the firewall. |
| 09:13.28 | Preatorian | The ip has been blocked completly in the firewall |
| 09:13.36 | Preatorian | so it doesnt even reaches the machine anymore |
| 09:13.41 | Preatorian | problem still persists tho |
| 09:17.39 | coppice | vlt: T.38 isn't much happier about packet loss than audio FAXing. I mostly helps with timing |
| 09:27.56 | *** join/#asterisk Da-Geek (~Da-Geek@213-193-154-130.perm.easynet.be) |
| 09:30.32 | vlt | Would this work: Put a second Asterisk with T.38 termination on the other side of the wifi connection, receive faxes there and scp the tiff files over wifi to the other asterisk machine to resend them (maybe could also be done with two hylfaxes with t38modem)? |
| 09:31.27 | Preatorian | jblack: got any other ideas? |
| 09:31.36 | jblack | shrug |
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| 09:40.12 | Preatorian | :) |
| 09:45.54 | pentanol | hi there |
| 09:46.52 | pentanol | does anybody knows how communicate rooms created on different asterisk server in the same number? |
| 09:50.22 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 09:51.58 | hurdman | pentanol: hi, can you give more details of your need ? |
| 09:56.43 | pentanol | hurdman of course, for instance... I use opensips as balancer between few asterisk servers and when client invite in the meeting room on the different asterisk server they can't hear each other contained in the some number room on another asterisk servers |
| 10:02.43 | hurdman | have you test an iax connection between your multiple servers ? |
| 10:11.38 | pentanol | hurdman no I've not had it, but I would? |
| 10:12.05 | pentanol | in case I use sip |
| 10:12.17 | *** join/#asterisk Gido-E (~gido@lounge.datux.nl) |
| 10:12.54 | pentanol | may I communicate asterisk over sip? I guessing I should call from one asterisk to another... where exist this room |
| 10:13.10 | kaldemar | ~book |
| 10:13.11 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 10:13.33 | pentanol | unhelpfull |
| 10:13.37 | kaldemar | pentanol: ^^ that book has examples on connecting two asterisk boxes with SIP or IAX2 |
| 10:13.52 | hurdman | pentanol: i use iax to dial between asterisk server to interconnect any type of call |
| 10:14.00 | pentanol | I knows how connect them via sip trunk |
| 10:14.12 | pentanol | it basicaly |
| 10:14.25 | pentanol | but it there room with the same number |
| 10:15.02 | pentanol | i'e' in created on one server and then on another one |
| 10:15.11 | pentanol | it* |
| 10:15.28 | hurdman | ( kaldemar old version of asterisk into the book, isn't it ? ) |
| 10:16.15 | kaldemar | hurdman: that's written for 1.4, but it's still valid for most parts. a newer edition is being written. |
| 10:16.47 | Gido-E | as long as there wil be a new one for 1.8 |
| 10:18.51 | hurdman | kaldemar: i'll buy it :D |
| 10:19.26 | kaldemar | you can download a free PDF if you wish. |
| 10:19.43 | Gido-E | Yep, but buying it is also a good thing. |
| 10:20.00 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 10:20.03 | hurdman | mmh i have to encourage free writing ;) |
| 10:21.11 | krion | free speech |
| 10:21.24 | pentanol | hm, I've that book |
| 10:21.56 | hurdman | where asterisk choose what number is shown in CALLERID(all) ? libpri or chan_dahdi or asterisk core source-something-else ? |
| 10:22.29 | Gido-E | hurdman it is a combination of id and name |
| 10:23.34 | hurdman | since an update of pri 4 month ago, i have the national number instead of the suscriber number shown /o\ |
| 10:23.35 | pentanol | but If I want to collect in database created rooms numbers and ip's of my asterisk nodes I should use AGI? no other way todo that? |
| 10:24.20 | hurdman | pentanol: or self made C module ;) |
| 10:24.50 | hurdman | ( seems to be a 3g patch erf ) |
| 10:26.39 | pentanol | how about cmd MYSQL? |
| 10:41.01 | *** join/#asterisk blaines (~blaines@173-14-197-193-huntsville.hfc.comcastbusiness.net) |
| 10:56.20 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 11:10.03 | pentanol | sounds itn's real to implement with cmd-mysql? |
| 11:12.26 | *** join/#asterisk Diogo (~diogoserr@bl6-198-208.dsl.telepac.pt) |
| 11:12.33 | pentanol | there only question about obtaining addr into exten config so as to inser it then in db, or make it static. |
| 11:13.11 | *** part/#asterisk Diogo (~diogoserr@bl6-198-208.dsl.telepac.pt) |
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| 12:09.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:11.28 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:12.44 | ariel_ | MOrning |
| 12:24.34 | hurdman | ariel_: hi |
| 12:25.01 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 12:25.04 | spenguin[work] | hey |
| 12:25.12 | hurdman | hi spenguin[work] |
| 12:25.34 | spenguin[work] | is there anything I should watch out for before I compile asterisk with a sangoma A200 card? |
| 12:36.23 | carrar | watch out for ninjas |
| 12:37.42 | ariel_ | or mugwats |
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| 12:52.42 | spenguin[work] | wokay |
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| 13:35.40 | *** join/#asterisk pigo (~7c7cf78c@gateway/web/freenode/x-sybxvfwxjkzjlcvj) |
| 13:35.47 | pigo | hi everybody |
| 13:36.36 | pigo | Can anybody suggest the easisest way to make IVR on asterisk (i mean which scripts to use: php/jsp/java/perl....) |
| 13:36.41 | *** join/#asterisk ManxPower-work (~manxpower@25.sub-75-234-99.myvzw.com) |
| 13:36.47 | ManxPower-work | ~answers |
| 13:36.48 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:36.55 | pigo | basically i need to store user recordings and play them back at some later point in time |
| 13:37.03 | pigo | with some browsing capability via IVR |
| 13:37.30 | ManxPower-work | pigo, I don't understand why you are coming here for VXML and other custom software support. |
| 13:37.53 | ManxPower-work | It's not like we are going to help you fix your JSP scripts |
| 13:38.00 | pigo | No No |
| 13:38.05 | pigo | i m not asking for that |
| 13:38.17 | carrar | pigo, write in vi |
| 13:38.21 | carrar | workse awesome! |
| 13:38.24 | pigo | i m just asking your opinion on which language would be best !! |
| 13:39.01 | pigo | @carrar ya dude i am using Gvim for last 3 years |
| 13:39.14 | carrar | then start writting |
| 13:39.54 | carrar | easiest way is to just type one out |
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| 13:40.46 | *** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 13:41.41 | kaldemar | pigo: you can do all that in extensions.conf. |
| 13:42.25 | pigo | but i think it is much easier if you write scripts and call them using AGI |
| 13:43.26 | pigo | i need to implement dynamic IVR, so i need to find out what language and which server would be supported best by asterisk's AGI |
| 13:43.30 | kaldemar | you should know what language you want to use. |
| 13:43.45 | carrar | use one you know |
| 13:43.48 | pigo | Actually i am new to all scripting languages |
| 13:43.50 | carrar | cause you have a lot of writting to do |
| 13:44.05 | pigo | and i tried some jsp code from internet but it is giving error |
| 13:44.15 | pigo | and i am having hard time debugging it |
| 13:44.21 | kaldemar | what makes you think it's easier with an AGI then? |
| 13:44.26 | pigo | (3-4 days) |
| 13:44.29 | pigo | haha |
| 13:44.49 | pigo | actually i need to implement dynamic IVR for users |
| 13:45.07 | pigo | which you anyway can not do directly in extension.conf dialplan |
| 13:45.13 | carrar | why not? |
| 13:45.44 | pigo | for example, i need to provide option to user to browse to and fro in his past recordings |
| 13:45.58 | pigo | how can i do that with something as static as a dialplan |
| 13:46.09 | pigo | i need it to be dynamically generated by some script |
| 13:46.22 | pigo | that generates vxml on the fly and submit it to asterisk |
| 13:46.24 | carrar | You could do that in a dialplan |
| 13:46.39 | pigo | @carrar can you tell me how ? |
| 13:46.42 | carrar | why do you need vxml? |
| 13:47.48 | pigo | @carrar OK, suppose i have to ask user "Press 1 to go to next audio snipplet" "Press 2 to go to prev audio snipplet" "Press 3 to play present audio snipplet" |
| 13:47.49 | carrar | you just said you don't need that ealier |
| 13:48.18 | kaldemar | pigo: that's still doable in extensions.conf. |
| 13:49.03 | pigo | ok, tell me how to do it for above case |
| 13:49.07 | carrar | no |
| 13:49.10 | carrar | teach yourself |
| 13:49.11 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:49.12 | carrar | thats the best way |
| 13:49.22 | carrar | you will learn more |
| 13:49.33 | pigo | that is fine |
| 13:49.39 | carrar | but it's doable |
| 13:49.43 | pigo | i m not asking you to do my work |
| 13:49.49 | carrar | sound slike you are |
| 13:49.54 | carrar | <pigo> ok, tell me how to do it for above case |
| 13:49.58 | pigo | but it is just that i m not getting any way in that direction |
| 13:50.06 | kaldemar | you can catch keypresses in dialplan and do stuff based on them. |
| 13:50.11 | pigo | may be my concept is wrong somewhere |
| 13:50.13 | carrar | ask a specific question |
| 13:50.41 | carrar | no one here is gonna spend time writting it for you unless you pay them |
| 13:50.58 | WIMPy | pigo: How would you do it in a script? Do the same thing. |
| 13:51.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:51.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.57 | carrar | draw it out in a flow chart and then tackle each section |
| 13:52.11 | carrar | You'll find it's easier then you think |
| 13:52.34 | pigo | http://lists.digium.com/pipermail/asterisk-users/2007-January/178744.html |
| 13:53.36 | carrar | You might want to lear perl (or insert language here) first if you are gonna write AGI |
| 13:54.59 | pigo | that i have done already, but leave that part |
| 13:55.05 | pigo | jsut focus on the link i gave |
| 13:55.12 | pigo | my question is exactly same as that |
| 13:55.13 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 13:55.30 | pigo | (i might have been unclear in starting, sorry about that) |
| 13:55.48 | ManxPower-work | Only you can decide which is better. |
| 13:56.03 | carrar | I believe your original question was: <pigo> Can anybody suggest the easisest way to make IVR on asterisk (i mean which scripts to use: php/jsp/java/perl....) |
| 13:56.15 | carrar | we said extensions.conf |
| 13:56.26 | ManxPower-work | I personally do ANYTHING that will involve a Dial in the dialplan. I use AGI scripts for non-dialplan type stuff like looking up userids in MYSQL |
| 13:57.08 | pigo | @carrar (i apologize for a misdirected question) |
| 13:57.14 | ManxPower-work | pigo, http://www.voip-info.org/wiki/view/Asterisk+tips+IVR+menu |
| 13:57.34 | carrar | I personally like perl for AGI stuff |
| 13:59.59 | ManxPower-work | carrar, I like the asterisk-perl AGI library. |
| 14:00.21 | ManxPower-work | pigo, Maybe if you READ THE ASTERISK BOOK you would not be asking so many questions |
| 14:00.26 | ManxPower-work | ~boook |
| 14:00.27 | carrar | the on in CPAN? |
| 14:00.28 | ManxPower-work | ~book |
| 14:00.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:00.29 | carrar | one |
| 14:00.45 | ManxPower-work | carrar, I download it from the author's site. |
| 14:00.59 | carrar | yeah, it works well |
| 14:02.37 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:04.27 | elred_ | that's great that you provide freely the book aswell |
| 14:04.31 | elred_ | tho i think to buy it |
| 14:10.33 | TheDavidFactor | Slightly OT: I've got a problem with a TCE400B from digium and they want the serial number to register the product for support on their website. The card is installed in a server that I have remote access to, does anyone know of a way to get the serial number remotely? |
| 14:10.39 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:10.55 | ManxPower-work | TheDavidFactor, No, |
| 14:11.28 | ManxPower-work | You'd think the serial number would be on the packing slip or something else in the package |
| 14:12.19 | TheDavidFactor | the only paperwork I have is the original invoice which doesn't have serial numbers on it |
| 14:13.00 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 14:13.26 | spenguin[work] | hey |
| 14:13.40 | spenguin[work] | whats the relation between wanpipe and sangoma cards |
| 14:14.18 | ManxPower-work | spenguin[work], wanpipe is the drivers for the sangoma cards |
| 14:14.29 | spenguin[work] | ok |
| 14:14.42 | spenguin[work] | well I installed the driver do I need the wanpipe router? |
| 14:14.44 | spenguin[work] | at boot |
| 14:14.57 | ManxPower-work | spenguin[work], did you pick TDM/Asterisk in the install? |
| 14:15.05 | ManxPower-work | (I don't remember the exact option name) |
| 14:15.23 | ManxPower-work | spenguin[work], because you should not have to do anything but follow the ./Setup promptes |
| 14:16.01 | spenguin[work] | I ran the /usr/sbin/wancfg_zaptel |
| 14:16.18 | spenguin[work] | I actually went the make src... way |
| 14:18.23 | ManxPower-work | "make src"> |
| 14:18.48 | ManxPower-work | Last I saw, the instructions say to run ./Setup, not make |
| 14:19.05 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 14:19.33 | spenguin[work] | ill try it out |
| 14:20.51 | ManxPower-work | spenguin[work], there's an entire document on Sangoma's wiki about using their stuff with Asterisk |
| 14:20.56 | ManxPower-work | or at least installing their stuff |
| 14:22.06 | ManxPower-work | spenguin[work], http://wiki.sangoma.com/wanpipe-linux-asterisk |
| 14:22.23 | spenguin[work] | thanks man |
| 14:22.37 | ManxPower-work | Usually I require dinner and drinks before holding someone's hand. |
| 14:25.10 | spenguin[work] | lol, well I was reading http://www.sangoma.fr/faq/faq_linux.htm |
| 14:25.30 | ManxPower-work | spenguin[work], but you're not "using linux" you're using Asterisk 8-) |
| 14:25.40 | spenguin[work] | yeah |
| 14:26.03 | ManxPower-work | spenguin[work], the ONLY drawback that I see with Sangoma is their much more complex setup process |
| 14:26.21 | spenguin[work] | I guess, seems interesting |
| 14:26.33 | spenguin[work] | complex == better control? |
| 14:28.51 | aceio | hi guys |
| 14:28.53 | ManxPower-work | spenguin[work], not in my opinion |
| 14:29.11 | aceio | so far now my dial plan is working |
| 14:29.23 | Kobaz | what's the preferred sip-t1 gateway |
| 14:29.36 | ManxPower-work | Kobaz, Asterisk, of course. |
| 14:30.01 | Kobaz | no sadly that's not going to work... this is the 5th t1 crash/problem we've had on a production box in 5 weeks, the customer is *pissed* |
| 14:30.13 | Kobaz | we need to move this off to a real gateway |
| 14:30.25 | aceio | i just wanted to ask if sip image p0s3-08-11-00 is a good image too use |
| 14:30.47 | Kobaz | i've already avoided analog cards for a while with great success |
| 14:30.55 | Kobaz | time to move to external t1s as well |
| 14:31.40 | Kobaz | ManxPower-work: if you actually looked at the code for libpri and etc, you would have the same lack of confidance that i'm having |
| 14:32.04 | ManxPower-work | Kobaz, and yet, we don't have the problems you have |
| 14:32.24 | Kobaz | ManxPower-work: you don't know that |
| 14:32.36 | ManxPower-work | Kobaz, I'd know about a crash |
| 14:32.45 | ManxPower-work | Kobaz, I take it you are using Digium cards? |
| 14:32.48 | Kobaz | sangoma |
| 14:33.00 | ManxPower-work | ah! At least you went with good hardware |
| 14:33.06 | Kobaz | i've had more problems with digium than anything else |
| 14:33.08 | Kobaz | rhino too |
| 14:33.19 | ManxPower-work | Kobaz, I don't recommend Digum cards anymore. Have not in years |
| 14:33.20 | Kobaz | yeah i think the hardware is fine |
| 14:33.42 | Kobaz | but the software driving the hardware is held together with shoe string and bubble gum |
| 14:34.06 | Kobaz | i've put in fixed for at least two libpri crashes so far |
| 14:34.11 | Kobaz | *fixes |
| 14:34.19 | Kobaz | now i got a kernel panic with dahdi |
| 14:34.22 | Kobaz | that's gonna be fun |
| 14:34.36 | ManxPower-work | Kobaz, we stopped having problems when we switched to Sangoma on our company PBX |
| 14:34.59 | ManxPower-work | no more echo, and no more problems with DTMF just stop working |
| 14:35.34 | Kobaz | yeah |
| 14:36.29 | Kobaz | the problem is |
| 14:36.37 | ManxPower-work | Kobaz, Digium people say "Have you tried our new cards?" To which I reply "Why would I use the same vendor I've had so many problems with?" |
| 14:36.37 | Kobaz | you can test this stuff in the lab for months and not hit a problem |
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| 14:37.28 | Kobaz | as soon as you put a box into production, that's when you see problems |
| 14:38.05 | seanbright | the only minor issue i ever had with wanrouter was that it would cause a kernel panic every time you ran 'wanrouter stop' |
| 14:38.12 | Kobaz | yeah |
| 14:38.21 | Kobaz | that's not a minor issue |
| 14:38.21 | seanbright | i say that tongue in cheek |
| 14:39.03 | Kobaz | the frusterating thing is yeah... i have four sangoma cards in a box that has an uptime for 400+ days |
| 14:39.13 | Kobaz | i have single t1s in other boxes that are running great |
| 14:39.24 | Kobaz | and all of a suddon you do a new install, and bang |
| 14:40.10 | Kobaz | i've seen plenty of dahdi kernel panics before, when playing with the proper combination of kernels, wanpipes, and dahdi to make things work |
| 14:40.13 | Kobaz | i'm tired of it |
| 14:40.29 | seanbright | and you're not going to take it anymore? |
| 14:40.33 | Kobaz | once you get it working, it's like okay... it seems to work... lets's run this for a while and hope it doesnt crash |
| 14:40.39 | Kobaz | heh, that's the idea |
| 14:40.40 | aceio | i just wanted to ask if sip image p0s3-08-11-00 is a good image too use |
| 14:40.51 | Kobaz | aceio: we have no idea what you're talking about |
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| 14:41.32 | Kobaz | i have a internal wiki page on which kernels work with which dahdi's which work with which wanpipes |
| 14:41.56 | seanbright | Kobaz: digium hardware? |
| 14:41.57 | ManxPower-work | loads chan_psychic.so and yells OMG THAT'S SICK! and unloads chan_psychic.so |
| 14:41.59 | Kobaz | sangoma |
| 14:42.02 | seanbright | ah |
| 14:42.25 | Kobaz | i was using digium, and had problems like what manx was talking about, dtmf echo, etc |
| 14:42.35 | Kobaz | and then switched to rhino since they were cheap |
| 14:42.46 | Kobaz | and had even more problems... and finally switched to sangoma, which has been the most solid |
| 14:42.54 | seanbright | p0s3-08-11-00 appears to be cisco firmware |
| 14:43.12 | seanbright | Kobaz: gotcha. sorry to hear that. |
| 14:43.15 | Kobaz | yeah |
| 14:43.32 | Kobaz | so, i think having the pbx not have a any line cards in it, will greatly reduce headaches |
| 14:43.42 | seanbright | i think digium offers OSS support |
| 14:43.56 | ManxPower-work | Kobaz, I've sold three Tellabs EC systems to connect to digium cards with hardware EC. 8-) |
| 14:43.59 | Kobaz | i'm kinda of tired compiling six different kernels to try and figure out which ones work with version x of dahdi |
| 14:44.23 | Kobaz | seanbright: i don't want support... i want it to "just work" (tm) |
| 14:44.34 | seanbright | oh |
| 14:44.35 | Kobaz | which is sangoma's slogan isn't it... heh |
| 14:44.37 | p3nguin | aceio: I prefer 8.11 on 7940/7960 phones. It's fine. |
| 14:44.45 | seanbright | "because it must work" |
| 14:44.55 | Kobaz | seanbright: yeah i know, it's oss, i know there are risks |
| 14:45.09 | Kobaz | many asterisk bugs i run into, i can fix myself... i have no problem with that |
| 14:45.26 | Kobaz | and asterisk itself has been pretty solid for me... it's all the outer layer stuff |
| 14:45.30 | ManxPower-work | We're stuck on 1.4.23.2 for several reasons |
| 14:45.59 | ManxPower-work | I really hope they fix the attended transfer issue in 1.4.x |
| 14:46.27 | ManxPower-work | at least later versions of 1.4.x |
| 14:46.38 | Kobaz | ManxPower-work: i would much prefer to stick to x.y.z for all production systems |
| 14:46.53 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
| 14:47.05 | Kobaz | but problem arrises when x.y.z+1 comes out and fixes a dozen bugs that you've run into... you upgrade.. but then you hit new bugs |
| 14:47.30 | Kobaz | heh |
| 14:47.51 | Kobaz | so in reality, i'm just not going to be able to truely have trust in a new version until i finish my unit testing suite |
| 14:48.04 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:48.09 | Kobaz | ...and i get the hardware out of the picture |
| 14:48.32 | Kobaz | because i don't have the expertise to rewrite libpri and dahdi |
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| 14:53.20 | p3nguin | What "attended transfer issue" exists in later 1.4 versions? I use later 1.4 versions and I'm wondering what I should be looking for. |
| 14:53.50 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 14:54.28 | ManxPower-work | p3nguin, at some (apparently random) time A specific attended transfer sequence results in Asterisk sending out the wrong SDP |
| 14:54.45 | ManxPower-work | once it start happening you have restart Asterisk |
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| 14:55.07 | ManxPower-work | If you don't do many consultative transfers I doubt you'd see the isseu |
| 14:55.08 | p3nguin | So if I do more attended transfers, I could run into it. |
| 14:59.12 | Slugs_ | morning |
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| 14:59.56 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 15:03.18 | Katty | hi |
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| 15:05.53 | ruben23 | Katty: hi |
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| 15:07.06 | Katty | hi ruben23 |
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| 15:25.25 | ChannelZ | yawns |
| 15:26.56 | ChannelZ | TGIF |
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| 15:35.16 | spenguin[work] | does asterisk need to be compiled with dahdi support, or ./configure without any parameters should suffice |
| 15:35.19 | spenguin[work] | ? |
| 15:36.10 | chazzam | if you want to use dahdi, then it needs to be compiled with dahdi support |
| 15:36.21 | spenguin[work] | ok |
| 15:36.34 | *** join/#asterisk Fubard (~brawr@173.10.235.205) |
| 15:36.37 | chazzam | configuring with no parameters should include it or leave it out depending on whether DAHDI is already installed |
| 15:37.07 | spenguin[work] | ok |
| 15:46.29 | spenguin[work] | http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=20 |
| 15:46.41 | spenguin[work] | whats the thing about redphone dahdi packages |
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| 16:01.31 | tzafrir_laptop | spenguin[work], that specific document is dated. |
| 16:01.54 | tzafrir_laptop | THat said, dahdi-{linux,tools} in svn trunk now include the redfone drivers |
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| 16:06.24 | afink | Hello everyone, I am having an issue forwarding long distance calls to a cell phone over a PRI. Here is a debug of the problem happening: http://pastebin.com/NyQuN93Z |
| 16:07.39 | afink | as soon as the remote end (cell phone) picks up the call the call is disconnected. Anyone have know anything I can try to solve this? |
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| 16:08.20 | *** join/#asterisk skrite (~skrite@pivotrac.texaspivot.com) |
| 16:10.19 | chazzam | afink, Asterisk is geing given teh disconnect message from the other end. I am not sure what the facility message is between the connect acknoledge and the disconnect, but we are sending the facility message |
| 16:10.27 | chazzam | s/we/asterisk/ |
| 16:10.32 | ManxPower-work | afink, set pridialplan=unknown and remove any other pri*dialplan options |
| 16:10.52 | ManxPower-work | then unload and load chan_dahdi.so |
| 16:11.20 | chazzam | wow I typo'd that well... |
| 16:11.35 | ManxPower-work | There is a reason the sample config file says you almost never need that option |
| 16:12.34 | afink | ManxPower-work: what file is that option in? |
| 16:13.08 | ManxPower-work | afink, chan_dahdi.conf |
| 16:13.24 | ManxPower-work | afink, unless you are using some crap GUI that puts stuff in different places |
| 16:13.36 | afink | no I'm not, thank you |
| 16:13.51 | ManxPower-work | or unless you ran the "automatic" dahdi config generator and don't know how to use it, then the option would also be in a different place |
| 16:15.02 | afink | I did run the automatic dahdi config generator |
| 16:15.23 | afink | I grepped for it though and couldn't find it anywhere |
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| 16:15.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:15.41 | spenguin[work] | Would you like to execute 'dahdi_cfg' each time wanrouter starts? |
| 16:15.45 | spenguin[work] | do I need that? |
| 16:16.20 | spenguin[work] | tzafrir_laptop: well I was just looking at it since the previous admin was working with those sources on the box |
| 16:16.26 | ManxPower-work | spenguin[work], I always answer yes. It's harmless to run it more than once. It will drop DAHDI calls, however, but since it's running on boot there should be no active calls |
| 16:16.38 | spenguin[work] | trudat |
| 16:16.48 | spenguin[work] | hands ManxPower-work a drink |
| 16:17.08 | Slugs_ | watched spenguin[work] spit in it ;/ |
| 16:17.28 | spenguin[work] | hey! its april fools - stop giving it away |
| 16:17.31 | tzafrir_laptop | spenguin[work], is your question about wanrouter or about redfone? |
| 16:18.04 | spenguin[work] | tzafrir_laptop: well I was replying to 19:24 tzafrir_laptop: spenguin[work], that specific document is dated. |
| 16:18.06 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 16:18.19 | spenguin[work] | but my most recent question was wanrouter related |
| 16:19.11 | tzafrir_laptop | ok. Because running dahdi_cfg should probably have an ill-effect on the redfone ones (I have no real idea, actually waiting for feedback from people there) |
| 16:19.42 | spenguin[work] | ok, I have no clue why the previous admin was playing around with redphone |
| 16:19.54 | spenguin[work] | it was just a week ago |
| 16:20.07 | spenguin[work] | Im guessing he was just following the howto blindly |
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| 16:20.38 | Slugs_ | been there.. |
| 16:20.49 | spenguin[work] | done that ? |
| 16:21.10 | Slugs_ | got the t-shirt |
| 16:21.11 | Slugs_ | ;) |
| 16:21.19 | spenguin[work] | hah |
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| 16:44.48 | *** join/#asterisk Fs0L (~Fs0L@136.223.19.19) |
| 16:45.13 | Fs0L | does anyone know if the old Flite application works with asterisk 1.4.30? |
| 16:46.45 | Fs0L | or was it replaced with festival? |
| 16:48.18 | skrite | hey all, with dahdi channels, how would i tweak the sensitivity of dtmf tones? i have mine way too sensitive |
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| 16:58.05 | hardwire | skrite: first you have to figure out what it's being sensative to. |
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| 16:58.26 | hardwire | and would changing the duration help? |
| 16:58.33 | hardwire | or would it break other things. |
| 16:58.46 | hardwire | if people are generating DTMF with their voices.. well.. you're hozed a little. |
| 16:58.54 | hardwire | turn features off when you don't need them enabled. |
| 16:59.21 | hardwire | I had to disable DTMF in meetme channels |
| 17:00.36 | skrite | ok, um.. maybe i have the wrong terminology. basic ivr setup, but sometimes when a button is pushed, like two becomes two-hundred-twenty-two |
| 17:00.40 | ManxPower-work | skrite, 1) remove relaxdtmg |
| 17:00.44 | ManxPower-work | ..er..relaxdtmf |
| 17:01.01 | ManxPower-work | yup! that sounds like classic relaxdtmf |
| 17:01.13 | skrite | ManxPower-work, thanks, will do |
| 17:01.26 | ManxPower-work | Sicne you didn't tell us what technology you are using, we can only make gueses |
| 17:01.55 | coppice | I wonder what is so broken about that relaxdtmf code |
| 17:02.21 | ManxPower-work | skrite, other things that can cause that is the gain being too high or too low. But since you have not told us what technology you are using, we can only guess. |
| 17:02.44 | ManxPower-work | coppice, the entire idea, as far as I can tell |
| 17:03.38 | coppice | if you relax the DTMF detector in spandsp in a similar way the false detect rate only slightly increases |
| 17:04.04 | ManxPower-work | coppice, what does it do in "relax" mode? |
| 17:04.27 | *** join/#asterisk mrbnet (~mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 17:04.38 | afink | ManxPower-work: I tried the pridialplan=unknown in chan_dahdi.conf and got the same results |
| 17:04.42 | coppice | it accepts a wider range of twist |
| 17:05.05 | afink | * with the long distance call forwarding |
| 17:05.07 | skrite | ManxPower-work, i am using a pci-express card, i don't remember the model, but is basically the tdm400 in pci-express form. dahdi service with asterisk on ubuntu 9.10 server |
| 17:05.13 | ManxPower-work | coppice, it appears that digium's just screws with the allowed gap in digits |
| 17:05.43 | ManxPower-work | skrite, *nod* Then the removing of relaxdtmf might help, but I suspect your gains are just out of whack |
| 17:05.46 | coppice | as far as I know they also just relax the twist |
| 17:05.58 | ManxPower-work | coppice, well the EFFECT is doubled or missing digits |
| 17:06.05 | skrite | ManxPower-work, i see the settings for rxgain and txgain both = 0.0 relaxdtmf=yes is commented |
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| 17:06.46 | coppice | well, the whole DTMF decoder in * has gone downhill over the years |
| 17:06.52 | ManxPower-work | skrite, then lower both by 2, then try 4 you should just have to reload chan_dahdi rather than unload/load it to apply the changes |
| 17:07.00 | afink | Here is another call trace of a failed call: http://pastebin.com/JqRGDm0F, It seems as though asterisk is trying to pass the call on rather than keep the channel open. |
| 17:07.09 | ManxPower-work | if that does not work then increase both by 2, then 4 |
| 17:07.13 | mrbnet | I have a polycom behind a nat which receives inbound call but cannot answer them. Does this sound like NAT issues or config issues? |
| 17:07.25 | skrite | ManxPower-work, so i can have negative values in there? |
| 17:07.30 | ManxPower-work | skrite, yes. |
| 17:07.36 | skrite | ok, will do. |
| 17:07.38 | skrite | thanks |
| 17:07.49 | ManxPower-work | Docs conflict about if you have to use whole numbers of can use something like 3.5 |
| 17:08.01 | ManxPower-work | I stick to whole numbers |
| 17:08.02 | skrite | ok |
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| 17:15.04 | ManxPower-work | mrbnet, nat issue |
| 17:15.05 | freezey | hey i a fax machine on one of the extensions i created. The fax can call out and recieve calls but how do i make it to faxes? is their a setting i am missing? |
| 17:15.30 | ManxPower-work | freezey, your question makes no sense. |
| 17:15.46 | freezey | ok |
| 17:15.49 | freezey | let me redo that then |
| 17:16.46 | freezey | i created a sip extension inside of asterisk.. i put a fax machine on the other end of an extension but the fax machine does not receive faxes or send |
| 17:16.54 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:17.43 | ManxPower-work | freezey, that's not suprizing. FaxOverVoiceOverIP tends not to work very well. |
| 17:18.05 | ManxPower-work | BTW, which Fax maxhine do you have that supports SIP? |
| 17:18.12 | *** part/#asterisk Fs0L (~Fs0L@136.223.19.19) |
| 17:18.27 | hardwire | FoVoIPoF |
| 17:18.38 | chazzam | I believe it is normally just termed foip |
| 17:18.53 | ManxPower-work | chazzam, But it is NOT Fax over IP. |
| 17:18.53 | hardwire | chazzam: I believe it deserves more letters than that. |
| 17:19.10 | hardwire | uses hfax protocol.. that's more foip than favoip would be |
| 17:19.12 | ManxPower-work | You are running a FAX over an RTP/SIP |
| 17:19.23 | ManxPower-work | as a voice codec. |
| 17:19.29 | chazzam | not if you do T.38... |
| 17:19.32 | e4voip | I gave my Ipod Touch to my 3 year old.... That thing is worthless |
| 17:19.37 | e4voip | she likes it |
| 17:19.41 | ManxPower-work | chazzam, none of us except you are talking T.38 |
| 17:19.44 | hardwire | anybody used gridengine? |
| 17:19.51 | e4voip | LOL worng chan |
| 17:20.02 | chazzam | weeee, my bad |
| 17:20.02 | freezey | ManxPower-work: it is some cannon printer/fax/scanner thing |
| 17:20.42 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:20.48 | ManxPower-work | freezey, and it supports SIP using ulaw? |
| 17:21.09 | ManxPower-work | I bet the fax machine IS a T.38 box. There's a total waste of money |
| 17:22.00 | freezey | yeah its a crappy fax |
| 17:22.08 | freezey | how can i get this dam thing to work normally |
| 17:22.45 | ManxPower-work | freezey, I doubt you ever will. But you might improve your chances if you knew what protocols the fax machine ACTUALLY support for Fax |
| 17:23.16 | freezey | yup |
| 17:23.29 | ManxPower-work | come back when you have that info |
| 17:23.38 | freezey | sure |
| 17:23.55 | ManxPower-work | then we might be able to tell you what might improve your chances. |
| 17:26.18 | e4voip | Hi -- Looking for this file -- catalog.xml |
| 17:26.24 | e4voip | Can someone tell me the path? |
| 17:27.16 | ManxPower-work | e4voip, I recommend "some other channe" since I doubt anyone here has any idea what you are talking about |
| 17:28.03 | e4voip | hmm.. Sadly keep hitting the incorrect chan |
| 17:28.13 | *** join/#asterisk Alagar (~Administr@122.164.42.35) |
| 17:28.17 | Qwell | e4voip: The answer is bacon. |
| 17:30.29 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:32.43 | beek | e4voip: You can always try the 'locate' command if this is on a Linux box. |
| 17:33.34 | e4voip | Hey beek.. Sadly this is a webserver I dont have SSH access to from this network.. |
| 17:33.43 | beek | yuck. |
| 17:34.31 | e4voip | Actually meant for the #magento channel |
| 17:34.35 | *** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net) |
| 17:35.38 | iratik | Where do include the context "detect" in this dialplan to test out AMD techniques? http://www.pastie.org/900577 ? |
| 17:45.03 | *** join/#asterisk e4voip (~chatzilla@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 17:48.27 | *** join/#asterisk blaines (~blaines@12.175.45.195) |
| 17:48.57 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 17:53.21 | *** join/#asterisk blaines (~blaines@12.175.45.195) |
| 17:55.21 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 17:55.24 | timeshell | Greetings |
| 17:56.06 | timeshell | I have H263 enabled on my users and in the SIP settings in *. However, using XLite 4 Beta, I cannot seem to get the video to go between 2 extensions. |
| 17:56.13 | timeshell | Any ideas? |
| 17:58.01 | *** join/#asterisk githogori (githogori@55.sub-75-210-92.myvzw.com) |
| 17:58.47 | timeshell | core show codecs shows that H263, H264+, H264 are present. |
| 18:01.56 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:03.37 | beek | Hello jaytee |
| 18:05.22 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:05.42 | Qwell | timeshell: No, it very much doesn't show that it's present. |
| 18:05.43 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 18:07.47 | *** join/#asterisk diegomad (mad@186.0.4.182) |
| 18:07.59 | jaytee | hi beek |
| 18:08.39 | timeshell | Qwell : Yes, I realized that after I said that |
| 18:08.39 | beek | jaytee: How goes the new gig? |
| 18:09.09 | timeshell | However, it is definitely enable in sip.conf and the extensions have the codecs assigned to them |
| 18:09.18 | Qwell | and you enabled support for video? |
| 18:09.24 | timeshell | videosupport=yes |
| 18:09.38 | Qwell | and the SIP debug says? |
| 18:09.40 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 18:10.48 | timeshell | Ack... how do I enable debug? I have it set to 10 and verbose 10, but it's not showing anything |
| 18:10.50 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 18:11.18 | doctorray | anyone here have polycom microbrowser experience? |
| 18:12.26 | doctorray | I'm looking to find/implement a "nowrap" equivelant on an IP450 since the text is so goddamn huge it can't display much more than 20-25 characters with variable width |
| 18:13.27 | *** join/#asterisk catlynx (sione@ocs.net) |
| 18:14.09 | catlynx | geez it took me so long to get on IRC cuz i forgot my password, I forgot what I was going to ask here :) |
| 18:15.31 | catlynx | oh i remeber now! |
| 18:15.54 | timeshell | Would video traffic go on ports other than the RTP ports? |
| 18:16.13 | catlynx | is there a way on a dial plan, after a caller hangs up the call not to exit the plan but go to the next step to execute a script to report some info of the call? |
| 18:16.26 | timeshell | If so, I may not have opened the firewall to it |
| 18:17.15 | iratik | Is there a different way I need to ask my question to get a response? |
| 18:17.40 | catlynx | timeshell: I would that all depends on your client |
| 18:17.59 | catlynx | timeshell: err! I would think that would depend on your client. |
| 18:18.02 | timeshell | catlynx : Also depends on the server. |
| 18:18.17 | Gugge | iratik: i have no clue what you are asking about |
| 18:18.21 | timeshell | If the blocks on the server are blocked, passthrough may not work |
| 18:18.28 | iratik | Where do include the context "detect" in this dialplan to test out AMD techniques? http://www.pastie.org/900577 ? |
| 18:18.42 | timeshell | I'm disabling the firewall on my server to test |
| 18:19.04 | Gugge | iratik: i saw the question before, i have no clue what you want. |
| 18:19.48 | timeshell | that seems to be part of it as I now see video debug messages |
| 18:21.48 | catlynx | timeshell: some firewalls would need a ALG to open some streaming ports, unelss you can get the client and server to use a define port that you can just open those ports on the firewall. |
| 18:21.56 | iratik | I'm trying to test out AMD detection , I am dialing using the context "outbound-long-distance". I can dial out fine .. I want to implement AMD (Answering Machine Detection) . I found a wiki article and found some dialplan samples to implement this. I placed the code in [detect]. The code I found does not contain a Dial command.. I don't know how to make the actions in [detect] apply when the call is dialed in [outbound-long-distance] |
| 18:23.14 | Gugge | iratik: what wiki article? |
| 18:23.17 | timeshell | catlynx I'm not sure where you're going with this. My asterisk server had a firewall which only allowed RTP ports 10000-20000, the normal audio ports in and out. I didn't think of video when I enabled my firewall on the server. If video uses different ports to pass through the server, I need to make sure those ports on the server are open. |
| 18:24.16 | iratik | Gugge: http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect |
| 18:25.29 | catlynx | timeshell: ok so you did define ports :) |
| 18:26.29 | catlynx | timeshell: could it be the 2 ends are not using proxy and trying to connect to each other with out the asterisk in the middle to send the stream? |
| 18:27.01 | timeshell | If that were the case, I believe it should be working since both devices are on the same network |
| 18:27.19 | timeshell | And the server is on a different subnet within the same network. |
| 18:27.38 | catlynx | timeshell: hmm just trying to toss you some ideas here |
| 18:27.39 | timeshell | There should be no natting between any of it. |
| 18:27.58 | timeshell | I understand. I'm just advising of the configuration |
| 18:28.42 | *** join/#asterisk blaines (~blaines@28.sub-75-235-240.myvzw.com) |
| 18:28.43 | timeshell | debug shows the SIP channel processing media-level (video) with H264 |
| 18:28.48 | timeshell | Sorry, 263... typo |
| 18:30.11 | timeshell | and then there's a messhttp://www.pastebin.ca/1855138 |
| 18:30.28 | iratik | Gugge: I guess i didn't make any sense still? |
| 18:32.14 | timeshell | Qwell: http://www.pastebin.ca/1855138 |
| 18:33.19 | Katty | hi |
| 18:33.30 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 18:33.32 | Qwell | it's Katty! HIDE! |
| 18:33.35 | iratik | hi! |
| 18:33.41 | Gugge | iratik: i guess you need to place it in the dial command somehow, but i dont know |
| 18:33.45 | Katty | :< |
| 18:33.49 | iratik | Gugge: :< |
| 18:33.53 | iratik | Thank you Gugge |
| 18:33.54 | Katty | sobs |
| 18:34.21 | iratik | Where is that [TK]D-Fender guy... he would always make me feel retarded for being alive.. but he would always know the answer |
| 18:34.36 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 18:34.43 | catlynx | any one know if you can make it so when a caller hangs up, to exceute the next step on a dial plan insted of exiting it so I can excute a script when the call ends to report some info? |
| 18:34.45 | moos3 | how can i repeat a context ? |
| 18:35.38 | Katty | goto(context,s,whatever) |
| 18:36.43 | moos3 | exten => s,n,Goto(context,s,whatever)? |
| 18:36.52 | Katty | yep |
| 18:36.58 | moos3 | ok thanks |
| 18:37.01 | Katty | mhmm |
| 18:37.36 | catlynx | guess what i want to do, cant be done :) |
| 18:37.58 | Katty | catlynx: h |
| 18:38.05 | Katty | infobot: asterisk standard extensions |
| 18:38.06 | infobot | somebody said asterisk standard extensions was http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 18:38.14 | Qwell | infobot: katty |
| 18:38.14 | infobot | katty is probably the only girl in the channel, so be nice to her |
| 18:38.20 | Qwell | lies |
| 18:38.21 | Katty | YEAH QWELL BE NICE |
| 18:38.30 | Qwell | nope |
| 18:38.32 | Katty | k |
| 18:39.02 | beek | Hello katty! |
| 18:39.09 | Katty | hi beekers |
| 18:39.11 | Katty | hugs beek |
| 18:39.18 | beek | hugs katty |
| 18:40.14 | catlynx | sweet! Thanks Katty that is what I needed :) |
| 18:41.47 | catlynx | time for lunch :) |
| 18:44.41 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 18:48.27 | *** join/#asterisk geneticx_wrk (~geneticx_@c-65-34-240-65.hsd1.fl.comcast.net) |
| 18:51.02 | moos3 | with the directory is it possible to have it read the name with festival instead of reading it out letter by letter? |
| 18:52.14 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 18:55.04 | *** join/#asterisk paulc (~Paul@unaffiliated/paulc) |
| 18:55.24 | *** join/#asterisk TonyOlivero (~ad9o@static-76-161-244-43.dsl.cavtel.net) |
| 18:55.45 | TonyOlivero | Hi all |
| 18:56.46 | TonyOlivero | If I'm logged into the Asterisk console, I see that remote connections are being issued,but is there a way I can see what commands are being sent? I'm trying to trace the behavior of a PHP script and it looks like it's loging in, but I can't tell why the commands aren't issuing properly. |
| 19:00.27 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 19:00.49 | iratik | ahi debug? |
| 19:00.53 | iratik | err.. agi debug? |
| 19:01.40 | TonyOlivero | I'll give that a shot |
| 19:05.09 | iratik | So .. I don't know if i should try my question again. I can't figure out how to apply the AMD/BackgroundDetect samples in the wiki into a real dialplan. They all have these commands in a context with extension s. But i though s is for outbound .. Also, there is no dial command... How can you do anything without a dial command? :: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 19:05.33 | iratik | Can anyone please help me out and illuminate this for me ... ? |
| 19:06.17 | *** join/#asterisk lynxsys (~lynxsys@213-152-36-251.dsl.eclipse.net.uk) |
| 19:06.20 | iratik | (I meant "inbound" in that last question where I discussed the usage of the "s" extension) |
| 19:06.50 | paulc | iratik: what are you trying to do exactly? |
| 19:06.56 | Fubard | I know asterisk is very customizable but does anyone know of a default script/setting that would reboot/resync the phones during off hours? |
| 19:07.00 | iratik | Just detect an answering machine ... |
| 19:07.41 | iratik | I'm calling my cell phone using the 41(my cell phone number) extension. I want the script to play "sorry" to the calling party when my voicemail is detected at the other end |
| 19:08.44 | paulc | ..or you could script it so you hit a key when you answer your cell... voicemail = no key pressed = deal with your caller appropriately |
| 19:09.20 | paulc | reading the wki page, looks like you want to outdial, pass the call to AMD after answering, then look at AMDSTATUS to see what the outcome was |
| 19:09.31 | iratik | Yes.. we are getting somewhere |
| 19:09.35 | iratik | How do i pass the call to AMD? |
| 19:09.35 | *** join/#asterisk Alagar (~Administr@122.164.42.35) |
| 19:09.51 | paulc | the tricky bit here is that you want the call to come from an inbound call, rather than making an outbound call |
| 19:10.00 | iratik | "Come from an inbound call" |
| 19:10.01 | iratik | no |
| 19:10.25 | paulc | You said you were dialing 41, an internal extension number, dialed to reach your cell? or did I misunderstand? |
| 19:10.40 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:11.15 | iratik | I'm making an outbound call... I'm dialing from extension 512, i type in 9417849.... The outbound long distance context is matched... Dial is executed , my cell phone rings.. I let it go to voicemail. I want that "detect" conext to realize that |
| 19:12.44 | iratik | I know the dial command is in [outbound-long-distance]. AMD is in [detect]... How do i connect the 2? |
| 19:14.14 | iratik | Sorry ... in my real dialplan I'm using 41.. in the dialplan i'm showing you .. its 9 .. becaise my real dialplan would be way too long .. What i'm not underestanding is more of a concept issue anyway |
| 19:15.12 | paulc | Take a look at the M option of Dial command - it allows you to run a macro once the called party has answered |
| 19:15.19 | iratik | Rats |
| 19:15.23 | iratik | I tried M and G |
| 19:15.38 | *** join/#asterisk fofware (~chatzilla@190.226.221.182) |
| 19:15.47 | iratik | I'm getting the feeling that you are doing exactly what i've been doing... reading asterisk bible.. reading the wikis and dial documentation |
| 19:15.55 | iratik | I'm not getting anywhere |
| 19:16.16 | paulc | What didyou do with the M option? It seems that that would work |
| 19:16.24 | TonyOlivero | rats, AGI debug doesn't seem to output AMI command unless I set something wrong. |
| 19:16.44 | paulc | Forget AMD for a minute - make a macro that plays a file, plays another file, then returns.. and test that... your ideal outcome is that it calls your cell, plays 2 prompts after you answer, then bridges the call to the original caller |
| 19:17.02 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.02 | paulc | if that works, you've proved that Dial with the M option works... then rejig the macro to use AMD |
| 19:17.42 | paulc | or.. you could do something clever with queues and agents - there's an option there that requires the "agent" to press a DTMF key to accept the call |
| 19:17.43 | iratik | paulc: I like that ... yeah.. I thought in all asterisk's hackery wizardry. AMD was somehow special and somehow magically injected itself into the plan and obtained the number through ESP or something . |
| 19:18.05 | iratik | I've read the dial options up and down |
| 19:18.28 | *** join/#asterisk geneticx_wrk (~geneticx_@c-65-34-240-65.hsd1.fl.comcast.net) |
| 19:19.08 | paulc | AMD is usually one end of a channel connection.. the other end is usually an outbound call.. then based on the result, you change "that end" to do something else (play a prompt, connect to an agent, whatever) |
| 19:19.10 | WIMPy | Erm, there was a direct option to dial that requires a keypress to ansert the call, wasn't there? |
| 19:19.38 | paulc | in your situation, you've already got 2 ends - the caller, and the answered call to the cell phone.. you need to find away to jump your connections around a bit.. |
| 19:20.27 | WIMPy | Maybe a combination of originate and bridge would be another way? |
| 19:20.37 | paulc | WIMPy: Hmm was there? sounds familiar but I don't see it in the Dial docs.. I think that was more queue/agent related? |
| 19:21.14 | paulc | Yeah - originate and bridge/transfer.. dump the caller in a conference, make an outbound call, run AMD, then transfer to the conference if ok, or redirect the existing call in the conference to somewhere else on failure |
| 19:22.48 | *** join/#asterisk qxork (~qxork@ip70-171-7-177.ga.at.cox.net) |
| 19:24.04 | Freeaqingme | I'm looking for some ATA devices that support 802.1X, any ideas? |
| 19:24.29 | WIMPy | Hmm. can;t find that option, either. Has it vanished? I've never used queues, so I don't think I have seen it there. |
| 19:24.45 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:39.18 | *** join/#asterisk micols (~mio@cl-85.cph-01.dk.sixxs.net) |
| 19:41.02 | *** join/#asterisk Circlefusion (~circlefus@adsl-80-74-57.sdf.bellsouth.net) |
| 19:46.25 | iratik | I feel like i know absolutely nothing about asterisk |
| 19:46.30 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:46.45 | Qwell | ~book |
| 19:46.46 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:46.49 | iratik | I can't even get this basic "Play hello world to the callEE" working ... http://www.pastie.org/900795 |
| 19:46.53 | Qwell | iratik: there's your solution |
| 19:46.57 | iratik | I've got that book open |
| 19:47.05 | iratik | I've read the dial command entry up and down |
| 19:47.13 | iratik | I read through basic dial plan construction up and down... |
| 19:47.15 | iratik | I've been on the wiki |
| 19:48.23 | iratik | I want to learn more about how the Dial command really works... M(hello) should send the call upon connection to [hello] .. Playback(hello-world) should then play it back to the person called |
| 19:48.49 | iratik | I don't get _any_ noops .. (With verbose at 9) ... Which means that the way asterisk is following this dialplan is much more different than I expect |
| 19:49.50 | ManxPower-work | iratik, Do you have Noop(...) in your Dialplan? |
| 19:50.03 | iratik | Did you even look at the pastie?!??????????????? |
| 19:50.19 | ManxPower-work | it had scrolled off my screen |
| 19:50.35 | iratik | http://www.pastie.org/900795 |
| 19:51.35 | ManxPower-work | [macro-hello] pay more attention to the Macro examples in the extensions.conf.sample |
| 19:52.27 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 19:52.47 | paulc | iratik: use context,extension,priority in your M command and macro definition |
| 19:52.52 | qxork | iratik... start small |
| 19:52.58 | qxork | don't use a macro first |
| 19:53.16 | qxork | tri just one context and grow from there. baby steps... baby steps |
| 19:53.42 | iratik | Omg I am going to explode You said "use context,extension,priority in your M command" |
| 19:53.59 | iratik | The definition of the M option in the Dial section in the asterisk bible makes no mention of those arguments |
| 19:54.07 | iratik | It just says M( x [ ^arg]) |
| 19:54.24 | qxork | iratik... relax. you are asking for help but coming across with an attitude |
| 19:54.45 | qxork | what is the cli output from the call? |
| 19:54.48 | iratik | qxork: My sincere apologies.... don't mean to direct to those trying to help me |
| 19:55.36 | qxork | do a test call, and paste the output from the cli... keep the verbose at 3 or 4 or even higher. such as, core set verbose 4 |
| 19:56.02 | iratik | I really need a development asterisk box |
| 19:56.08 | iratik | the production traffic is insane |
| 19:56.22 | paulc | uh.. YES! |
| 19:56.32 | paulc | even running on a VM or something |
| 19:57.55 | iratik | I'm using a macro because I'm under the impression that its the only way to execute commands while the call is active.. If i put commands after Dial.. the only get executed when the call is over (and only with the g option) |
| 19:58.12 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 19:58.54 | iratik | I guess the big black-hole that I can't seem to fill with knowledge is "where do you put the commands you want to be executed once the destination answers"? |
| 20:01.25 | qxork | iratik... without seeing anything... I'll just say this: 1. name all macros macro-xxx so hello would be macro-hello. then use a standard scheme of s,1,xxx s,n,yyyy etc. |
| 20:05.52 | iratik | I'm starting to suspect that maybe i'm not incompetent.. maybe its this freepbx junk interferring with my testing |
| 20:06.07 | iratik | http://pastie.org/900824 |
| 20:06.21 | *** join/#asterisk CyberPony (~CyberPony@66-194-25-11.static.twtelecom.net) |
| 20:06.39 | iratik | Here is the dial plan |
| 20:06.46 | iratik | http://pastie.org/900825 |
| 20:07.40 | qxork | exten => _41. should not be. use s |
| 20:08.04 | *** join/#asterisk sahafeez (~sahafeez@99-40-7-14.lightspeed.sntcca.sbcglobal.net) |
| 20:08.08 | qxork | on the macro |
| 20:08.19 | iratik | I'm not even calling that macro |
| 20:08.35 | iratik | You guys said to do everything in one context then i could split it up later |
| 20:09.38 | qxork | What are you actually trying to do? |
| 20:10.20 | qxork | you're playing a background without an answer... which normally will trigger an answer, but isn't best practice |
| 20:10.55 | iratik | what i'm really trying to do is apply the "BackgroundDetect" example in the wiki |
| 20:11.32 | iratik | So .. I don't know if i should try my question again. I can't figure out how to apply the http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 20:11.57 | qxork | this is an inbound call... |
| 20:12.00 | *** join/#asterisk Alagar (~Administr@122.164.42.35) |
| 20:12.10 | qxork | normally amd would be for outbound calls |
| 20:12.21 | iratik | The example in the wiki is for an inbound call? |
| 20:12.51 | iratik | I was wondering why they were using "s" ... which is usually used for inbound calls ... I thought i wasn't reading the manual closely enough |
| 20:13.52 | iratik | All I'm trying to do is apply AMD to a Dial command .. When I place a command immediately after dial.. its not getting executed until the call ends .. Until i figure out how to execute commands during the call .. i'll never be able to use AMD right? |
| 20:14.25 | iratik | I'm working this all out with Playback() ... because it gives me audible feedback on both sides of the connection that the command has executed |
| 20:14.26 | qxork | the wiki shows the examples in outbound |
| 20:14.41 | iratik | They all use s! |
| 20:14.44 | iratik | And no dial command |
| 20:15.39 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 20:15.39 | iratik | I wish someone would just explain to me the part i'm really really confused about is that.. I realize i'm missing something fundamental .. can someone just please explain it to me and put me out of my misery? |
| 20:16.03 | ManxPower-work | iratik, AMD is normally used for automatic calls and those work differently |
| 20:16.40 | iratik | wait... so i should be testing this out with AMI Originate and passing the [outbound] context to get this to work? |
| 20:16.48 | ManxPower-work | I doubt you'll be able to use AMD in regular dialplan. I've only ever heard of it being used with manager or .call files. |
| 20:17.16 | iratik | I was just testing it in the dialplan to remove any variables the AMI might introduce |
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| 20:21.47 | Wildy | мÑÑв =) |
| 20:22.21 | Wildy | oops sorry |
| 20:22.37 | Wildy | it seems /amsg can broadcast to other networks :( |
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| 20:33.28 | rafuch0 | Can Asterisk work with a Rockwell International HCF 56k Data/Fax/Voice/Spkp (w/Handset) Modem ? |
| 20:35.22 | Qwell | rafuch0: Sure, if you write drivers for it. |
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| 20:46.23 | p3nguin | Sometimes I get a CALLERID(num) value that has +1 on the front of the 10-digit phone number (but not always (depends on where the call originates)), and I would like to know how to strip it off when it does exist. Can someone give me a hint or a shove in the right direction? My first thought was to just shave off the first two characters, but since it doesn't always happen, I want to make it conditional. |
| 20:47.42 | Maliuta | p3nguin: you are aware that +1 is the correct format for the number? |
| 20:47.57 | Maliuta | p3nguin: if I were to call you it would be +61 |
| 20:48.12 | p3nguin | Yes, but since we don't use +1 in the US, I need to remove it to make the number compatible with our PSTN. |
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| 20:48.44 | ManxPower-work | p3nguin, + means "when calling this number use your country specific prefix for dialing international calls if needed. |
| 20:48.48 | p3nguin | we just use 1 when dialing long distance. |
| 20:49.00 | ManxPower-work | p3nguin, + only applies to international calls |
| 20:49.17 | p3nguin | Great, a lecture on something I didn't ask about. |
| 20:49.24 | Maliuta | us -> canada is international |
| 20:49.27 | ManxPower-work | it can be in front of any number, but it only MEANS something if that number is in a different country from where you are |
| 20:49.57 | ManxPower-work | Maliuta, use canada and about 13 other (smallish) countiries use country code "1" |
| 20:50.02 | p3nguin | US -> Canada isn't international calling from my keypad's standpoint. |
| 20:50.33 | ManxPower-work | not use, USA |
| 20:50.51 | Maliuta | ManxPower-work: if I want to call the USA I have to use +1 |
| 20:51.01 | ManxPower-work | I think in much of europe "0" is the toll prefix" and "00" is internations. |
| 20:51.27 | ManxPower-work | Maliuta, no, you would use your country specific prefix for calling inernationally |
| 20:51.31 | WIMPy | p3nguin: execif, set and ${x:n} |
| 20:51.39 | ManxPower-work | what IS the dialing code for international call for your country? |
| 20:51.40 | p3nguin | If all my numbers had the +1 on it, I would do that. |
| 20:51.57 | p3nguin | Since some do not, I want to make it conditional. |
| 20:52.08 | ManxPower-work | p3nguin, exten => _+.,1,Goto(${EXTEN:1},1) |
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| 20:52.19 | ManxPower-work | with the correct ,n, priority, of course |
| 20:52.21 | Maliuta | using the international dialing prefix in all cases with the + should be the right thing to do, your carrier should be able to correctly charge regardless if the call is local, long distance or international |
| 20:52.42 | beek | p3nguin: Use the REGEX function for the conditional in execif if you really want to change the callerid. |
| 20:52.47 | p3nguin | I would rather not create another extension. There has to be a sane way to do it within the same exten as another priority. |
| 20:52.53 | ManxPower-work | Maliuta, where is the + on my telephone keypad? |
| 20:53.02 | WIMPy | Actually tehere are even carriers that don't know theit own country :-( |
| 20:53.12 | ManxPower-work | p3nguin, that IS the best way. |
| 20:53.36 | Maliuta | ManxPower-work: this is from a PBX perspective, not your dumb(as opposed to smart) phone |
| 20:53.45 | Maliuta | ManxPower-work: my mobile has a + key |
| 20:53.53 | WIMPy | Err, are we talking calling or called number? |
| 20:54.02 | beek | p3nguin: use execif, regex, set and ${x:n} if you want to do it on the same priority. |
| 20:56.58 | WIMPy | However I hven't found a good solution for that problem ragarding called numbers (user input). |
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| 21:08.05 | timeshell | Looking for assistance in finding out why video isn't working. Both servers have 1.6.0.26. videosupport=yes on both. Both have users assigned H263 and H263+ codecs. |
| 21:09.01 | timeshell | Current debug info here: http://www.pastebin.ca/1855277 |
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| 21:15.28 | p3nguin | beek: What do you think about this? ExecIf($[${REGEX("^+1" ${CALLERID(num)})} = 1]|Set|CALLERID(num)=${CALLERID(num):2}) |
| 21:16.27 | beek | p3nguin: What version of asterisk is this? The pipe symbols are deprecated. |
| 21:16.31 | p3nguin | 1.4 |
| 21:16.51 | beek | All I can say is "When in doubt, try it out." |
| 21:17.06 | beek | It looks as though you have the right flavor. |
| 21:17.07 | p3nguin | core show application ExecIf shows to use pipes, but I can change to commas if needed. |
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| 21:17.53 | WIMPy | A simple comparison instead of regex might be more efficient. |
| 21:17.59 | p3nguin | That would be nice if it works as is with no other tinkering. I'm not very swift with regular expressions. |
| 21:19.21 | p3nguin | func_strings.c:142 regex: Malformed input REGEX(): Invalid preceding regular expression |
| 21:19.24 | beek | p3nguin: Actually, the '+' sign isn't quite right. That's would mean "one or more" of the preceeding character. You'll want to escape it. |
| 21:19.37 | p3nguin | ast_expr2.fl:415 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 1 |
| 21:20.12 | beek | p3nguin: Give me an example of a phone number that you want to adulterate as it would appear in your CALLERID(num) field... |
| 21:20.17 | WIMPy | Just compare the first two digits to +1. |
| 21:20.39 | beek | Is the plus sign really a part of CALLERID(num)? |
| 21:20.52 | p3nguin | beek: +16183396611 |
| 21:21.12 | beek | WIMPy: I'm a heavy Perl user so I just think in terms of regex. |
| 21:21.20 | p3nguin | Yeah, it's really part of CALLERID(num). That's why I want to take it out when it shows up. |
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| 21:22.15 | beek | The regex would be: '^\+1'... you need to escape it. |
| 21:23.49 | p3nguin | It seems like the syntax error thing is where it's hanging. |
| 21:24.18 | p3nguin | unexpected '=', expecting $end; Input: = 1 and the ^ symbol points to that = before the 1. |
| 21:25.41 | p3nguin | ${REGEX("^\+1" ${CALLERID(num)})} should = 1 if ${CALLERID(num)} starts with +1, right? |
| 21:26.12 | beek | p3nguin: Forget the regex and if you're really seeing '+1' then do ${CALLERID(num):0:2} = '+1' |
| 21:26.48 | p3nguin | I don't understand what you mean. |
| 21:27.00 | p3nguin | On, yeah I do. |
| 21:27.14 | beek | Don't use regex... execif($[${CALLERID(num):0:2} = '+1']...) |
| 21:27.18 | p3nguin | Compare ${CALLERID(num):0:2} to +1 rather than dealing with regex. |
| 21:27.24 | beek | Yes |
| 21:27.37 | p3nguin | I was a little slow at first. :/ |
| 21:28.15 | beek | I'm outta here... I was just leaving when you IMed me. You have the idea... |
| 21:30.18 | p3nguin | That's much easier and works like a charm. |
| 21:30.23 | p3nguin | Thanks for the assistance. |
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| 21:31.27 | trentcreek | ~book |
| 21:31.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:37.25 | hardwire | erm.. so with sip registrations will the dnsmgr attempt to use the same IP for a named registration server every time. Is there a mechanism to use another registration server in the event of a timeout? |
| 21:37.47 | hardwire | if so.. I'll gladly add a local zone file for some services that expect me to use another registration server if their firewall decided to block me. |
| 21:37.57 | hardwire | sigh. |
| 21:38.44 | WIMPy | It cahnges randomly, no idea if there's any mechanism in case of a failure. Don't think so. |
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| 22:12.36 | iratik | http://www.pastie.org/901039 |
| 22:13.01 | iratik | My basic question is still where I need to put commands to execute upon the callee picking up the call. ? |
| 22:16.02 | paulc | iratik: you need to understand the channel endpoints |
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| 22:16.09 | paulc | your pastebin code will work fine if you originate the call |
| 22:16.20 | paulc | the question then is "how do I link that successful outbound call to my original caller" |
| 22:16.41 | paulc | options are clever stuff with M in the dial command (via a macro).. or something with conferencing and redirecting.. |
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| 22:17.02 | iratik | So trigger AMD with M(ansdetect) and [macro-ansdetect] ? |
| 22:17.11 | paulc | or decide that your time is worth more than this, and just go down the queues + agents + DTMF confirmation route, because that will give you exactly what you want, for the cost of "having to press a button when the call is answered" |
| 22:17.40 | iratik | If the called party has to press a button.. that will not do at all |
| 22:17.45 | paulc | Something like that.. not sure if you want M(ansdetect) or M(ansdetect,s,1) or similar.. but some experimentation will give you the answer |
| 22:18.02 | paulc | the other, simpler alternative is to set your Dial timeout to be a few seconds less than your "transfer to voicemail on no reply" timeout |
| 22:18.12 | paulc | telco timeout = 15 seconds.. set your Dial timeout to 12 or something |
| 22:18.25 | paulc | and assume you'll answer your phone quick enough if you're close to it |
| 22:19.29 | iratik | Thank you for being so patient with me :) |
| 22:19.41 | WIMPy | Won't help it you're out of coverage or simply switched off. |
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| 22:21.19 | paulc | WIMPy: true - which brings us back to the magic of AMD.. or just pressing a key.. |
| 22:21.39 | paulc | personally I think it might be nice to know who's calling me on my cell phone and give me the option of accepting or rejecting.. but that's just me. |
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| 22:22.26 | iratik | Holy sweetness it worked :D |
| 22:22.40 | iratik | The depth of my frustration is only exceeded by the joy of my triumph |
| 22:22.58 | WIMPy | Well, if you can set calling number that's ace. |
| 22:23.22 | WIMPy | iratik: Congratulations |
| 22:23.27 | WIMPy | So how did it work? |
| 22:23.42 | iratik | Well.. I learned a lot of things the documentation doesn't necessarily say |
| 22:23.58 | iratik | Mainly.. that AMD really only works with Originate or various auto-dialing approaches |
| 22:24.03 | WIMPy | I guess, we all know that part. |
| 22:24.19 | iratik | So testing it using regular dialplan just for testing ... was a dead end ... everything i tried there was in vain |
| 22:25.12 | WIMPy | BTW: Does anyone have an idea what I have screwed up so that core show application/function ... only shows empty templates? |
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| 22:28.09 | iratik | http://www.pastie.org/901039 ... Just in case you are curious.. stil doesn't really "work" .. but it gives me enough indications that its working enough for me to tweak it until it does |
| 22:28.24 | adam_g | hi... if ive got a custom agi script included in an extension, and it quickly forks and returns 0, what possible reasons would it still be blocking an extension excution? |
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| 22:32.54 | Kobaz | adam_g: do you understand how agi works? |
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| 22:36.08 | carrar | Morning!! |
| 22:36.20 | iratik | Morning!! |
| 22:36.45 | ttwhy | Hi, anyone tried to get a FAX with 1.6.2? My Fax get stored correctly (in spool dir) but i still get a error message (which prevent asterisk to make the next logical steps :/ ) WARNING[3639] app_fax.c: Error transmitting fax. result=49: The call dropped prematurely. |
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| 22:44.36 | hardwire | with permit=/deny= in sip.conf I could swear I could allow (via insecure=port,invite) a range of IP addresses IP based authentication into my * server. |
| 22:44.44 | hardwire | that may not be the case if a fromuser is set. |
| 22:44.55 | hardwire | from the remote hosts calling in/through * |
| 22:46.15 | adam_g | Kobaz, well, i thought i did. but apparently i do not :) |
| 22:46.41 | hardwire | like with broadvoice, I have their entire server range specified however the IP coming in is being sent a fake auth rejection |
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| 22:47.56 | QbY | can asterisk throw an SNMP trap when a peer/handset goes unreachable |
| 22:49.03 | paulc | QbY: I don't think so, but I'm not 100% sure |
| 22:49.25 | QbY | well does the asterisk snmp have much support for peers? |
| 22:49.32 | QbY | would like to load it into our nms |
| 22:49.41 | WIMPy | doesn't think so, either. But you can always listen on AMI. |
| 22:49.51 | QbY | true true |
| 22:50.56 | iratik | Most NMS status check / uptime detection systems support a module that checks sip connectivity on a port ... it would be arduous but you could apply this module to each extension |
| 22:51.07 | iratik | I know nagios has something like sip_check |
| 22:51.34 | WIMPy | it should be pretty straight forward to do smth that listens on AMI and then throws anmp traps on certain events if you need it that way. |
| 22:52.00 | QbY | Where can I find info on connecting/listening to AMI? |
| 22:52.02 | WIMPy | s/anmp/snmp/ |
| 22:53.05 | WIMPy | I found voip-info.org a good starting point, but always bear in mind that the infomation there may be inaccurate and outdated. |
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| 22:58.28 | manxpower | ~answers |
| 22:58.29 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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| 22:59.50 | hardwire | apparently permit is ignored if alwaysauthreject is enabled |
| 23:00.11 | hardwire | according to the source |
| 23:00.16 | hardwire | and.. now according to my testing too. |
| 23:01.12 | manxpower | hardwire: that makes sense. |
| 23:01.50 | hardwire | manxpower: I'd prefer a conditional permit/deny check first. |
| 23:02.03 | manxpower | write a sometimesauthreject 8-) |
| 23:02.14 | hardwire | alwaysauthrejectexceptsometimes |
| 23:02.34 | hardwire | alwaysauthrejectexceptwhenyouwanttobeanITSP |
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| 23:03.07 | manxpower | hardwire: alwaysauthreject seems to be designed to just not expose user account info. If the device is sending the right password it should not be a problem |
| 23:03.20 | hardwire | yar. I don't want to open a bug just to ask "Is this really what you wanted?" without a patch.. and I don't want to write a patch without knowing the answer. |
| 23:03.23 | hardwire | heh |
| 23:03.38 | hardwire | for IP based auth it gets in the way |
| 23:04.32 | hardwire | also.. I need to add support for querying a sip registration status from asterisk out. |
| 23:04.37 | hardwire | to tell me what IP address it used. |
| 23:05.04 | manxpower | hardwire: I suspect the original design was not intended to cover ip based auth. can you disable it per peer? |
| 23:05.11 | hardwire | manxpower: no |
| 23:05.35 | hardwire | permit should explicitely disable it if defined in a peer.. /me submits a patch |
| 23:05.59 | manxpower | I think you should just explicitly disable it per peer, not automatically. |
| 23:06.04 | hardwire | dealign with broadvoice isn't that fun atm |
| 23:06.34 | hardwire | they limit 10 valid registrations per 30 seconds per IP |
| 23:06.44 | hardwire | their expiry is 30 seconds |
| 23:07.01 | hardwire | so if I reload sip or if an automated process reloads sip.. sip registration is resent. |
| 23:07.08 | hardwire | and I'm banned for an hour |
| 23:07.21 | hardwire | the solution is to use a DNS SRV compliant endpoint that will go tothe next registration server |
| 23:07.31 | hardwire | as well as send calls to that server and expect them from that server |
| 23:08.20 | hardwire | my solution with asterisk, up until I realized they answered a call if you make a call through a a different server you are registered through, was to use DNS RR (local zone) and dnsmgr and let it cycle through |
| 23:08.38 | hardwire | and then allow all of those IP in via permit= |
| 23:08.43 | hardwire | so thats why all this came up |
| 23:09.06 | hardwire | instead.. I'm going to do things a little differently and check for sip registration retries in excess of 2 and rewrite /etc/hosts automatically. |
| 23:09.16 | chazzam | ask about the intention in asterisk-dev? |
| 23:09.24 | hardwire | chazzam: I'm not allowed. :) |
| 23:09.51 | hardwire | besides.. I'm a bit of a corner case ATM |
| 23:10.32 | hardwire | what weirds me out is BV has two identical priority servers in DNS-SRV |
| 23:10.41 | hardwire | so your endpoint really has to know what's going on. |
| 23:11.36 | hardwire | I attempted to use siproxd to rectify this, but it doesn't grok DNS-SRV |
| 23:11.43 | hardwire | I registered through it just fine however. |
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| 23:36.28 | grabes | Is there anyway to compile asterisk 32bit on a 64 bit system |
| 23:36.55 | hardwire | yup |
| 23:37.20 | hardwire | either set up a 32 bit chroot with all the dev tools or specify a make flag |
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| 23:37.41 | hardwire | -m32 |
| 23:38.00 | grabes | hardwire, thanks... any idea what the make flag would be on FreeBSD |
| 23:38.17 | hardwire | if it's gcc wouldn't it be the same? |
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| 23:43.33 | carrar | grabes: search away http://www.google.com/bsd |
| 23:44.38 | Qwell | why bother building 32-bit? |
| 23:47.30 | grabes | @Qwell, I have a binary module thats only provided for 32 bit |
| 23:48.06 | Qwell | grabes: Which one? |
| 23:48.28 | hardwire | a "binary" module.. no doubt |
| 23:48.49 | grabes | heh |
| 23:49.19 | grabes | Basically, I want to test the viability of us moving to a freebsd jail environment, and in order to do so I need to test g729 |
| 23:49.24 | grabes | unfortunately |
| 23:49.45 | Qwell | codec_g729a isn't only available as 32-bit |
| 23:50.13 | grabes | Qwell: For FreeBSD as well ? |
| 23:50.39 | Qwell | There are no OSes (supported or otherwise) that codec_g729a is only provided for 32-bit. |
| 23:51.05 | hardwire | grabes: what version of asterisk? |
| 23:51.24 | grabes | I am looking to run it on 1.6.1 |
| 23:51.35 | hardwire | grabes: erm.. so you're wanting to test the "free" ones. |
| 23:51.43 | Qwell | There are no 1.6.1 FreeBSD modules for codec_g729a. |
| 23:52.05 | grabes | 1.6.0, or 1.6.2 ? |
| 23:52.08 | hardwire | Qwell: there are at asterisk.hosting.lv |
| 23:52.10 | Qwell | No |
| 23:52.15 | Qwell | hardwire: No there aren't. |
| 23:52.23 | hardwire | ah.. for 1.6.0 only |
| 23:52.26 | grabes | Qwell: none of those FreeBSD codecs are for 64 but |
| 23:52.27 | Qwell | and nobody in here would help with them, if there were. :) |
| 23:53.04 | hardwire | Qwell: ask Eugene M. Zheganin |
| 23:53.08 | hardwire | err |
| 23:53.11 | hardwire | grabes: ^ |
| 23:53.18 | hardwire | maybe he'll tune you into the compiling procedure |
| 23:53.23 | hardwire | for x86_64 |
| 23:53.28 | Qwell | I recommend asking your attorney first. |
| 23:53.35 | hardwire | for testing.. meh. |
| 23:53.40 | grabes | hardwire: already have it has to do with the ipp and linux "emulation" under freebsd |
| 23:53.48 | hardwire | eww |
| 23:54.06 | grabes | My attorney says I am aloud to test all I want :) |
| 23:54.12 | Qwell | fire him. |
| 23:54.20 | Qwell | and this discussion stops now |
| 23:54.23 | hardwire | well.. I wonder why digifolk haven't compiled it yet. |
| 23:54.39 | Qwell | hardwire: nobody has asked for it |
| 23:54.42 | hardwire | cause it's there for 1.4.x in multiple flavors |
| 23:54.59 | hardwire | grabes: interested? |
| 23:55.24 | hardwire | 10 dolla codec codec!!.. fully supported long time.. |
| 23:55.56 | hardwire | my eyes feel funny |
| 23:56.00 | hardwire | stops touching them |
| 23:56.02 | *** part/#asterisk adam_g (~adam@static-173-50-141-22.ptldor.fios.verizon.net) |
| 23:56.18 | grabes | hardwire: who is it through? I have no problems paying |
| 23:56.31 | grabes | I haven't seen a commercial source for it |
| 23:56.32 | hardwire | digium |
| 23:56.51 | Qwell | grabes: Call Digium sales. Tell them you want X licenses to be used on FreeBSD with Asterisk 1.6.x. See what they'll do. |
| 23:57.02 | hardwire | X is a number. |
| 23:57.12 | hardwire | x is also a number. |
| 23:57.22 | hardwire | they don't have to be the same number. |
| 23:57.38 | Qwell | X has to be large enough to justify it for us. |
| 23:57.52 | hardwire | and x has to be <= 2 |
| 23:57.59 | carrar | 1,000,000 |
| 23:58.05 | grabes | haha |
| 23:59.45 | manxpower | Hopefully x will be >= 2 |
| 23:59.56 | hardwire | nono X |